Deprecate the static RtpRtcp::Create() method.
The method is being used externally to create instances of the deprecated internal implementation. Instead, I'm moving how we instantiate the internal implementation into the implementation itself and move towards keeping the interface separate from a single implementation. Change-Id: I743aa86dc4c812b545699c546c253c104719260e Bug: webrtc:11581 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176404 Commit-Queue: Tommi <tommi@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31420}
This commit is contained in:
parent
878808892d
commit
fae05624ec
@ -33,11 +33,11 @@
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#include "modules/pacing/packet_router.h"
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#include "modules/rtp_rtcp/include/receive_statistics.h"
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#include "modules/rtp_rtcp/include/remote_ntp_time_estimator.h"
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#include "modules/rtp_rtcp/include/rtp_rtcp.h"
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#include "modules/rtp_rtcp/source/absolute_capture_time_receiver.h"
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#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
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#include "modules/rtp_rtcp/source/rtp_packet_received.h"
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#include "modules/rtp_rtcp/source/rtp_rtcp_config.h"
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#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
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#include "modules/utility/include/process_thread.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/critical_section.h"
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@ -507,7 +507,7 @@ ChannelReceive::ChannelReceive(
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if (frame_transformer)
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InitFrameTransformerDelegate(std::move(frame_transformer));
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_rtpRtcpModule = RtpRtcp::Create(configuration);
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_rtpRtcpModule = ModuleRtpRtcpImpl2::Create(configuration);
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_rtpRtcpModule->SetSendingMediaStatus(false);
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_rtpRtcpModule->SetRemoteSSRC(remote_ssrc_);
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@ -29,6 +29,7 @@
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#include "modules/audio_coding/include/audio_coding_module.h"
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#include "modules/audio_processing/rms_level.h"
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#include "modules/pacing/packet_router.h"
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#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
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#include "modules/utility/include/process_thread.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/event.h"
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@ -530,7 +531,7 @@ ChannelSend::ChannelSend(
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configuration.local_media_ssrc = ssrc;
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_rtpRtcpModule = RtpRtcp::Create(configuration);
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_rtpRtcpModule = ModuleRtpRtcpImpl2::Create(configuration);
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_rtpRtcpModule->SetSendingMediaStatus(false);
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rtp_sender_audio_ = std::make_unique<RTPSenderAudio>(
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@ -16,6 +16,7 @@
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#include "api/audio_codecs/audio_format.h"
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#include "api/task_queue/task_queue_factory.h"
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#include "modules/rtp_rtcp/include/receive_statistics.h"
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#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
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#include "rtc_base/critical_section.h"
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#include "rtc_base/location.h"
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#include "rtc_base/logging.h"
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@ -51,7 +52,7 @@ AudioChannel::AudioChannel(
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rtp_config.outgoing_transport = transport;
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rtp_config.local_media_ssrc = local_ssrc;
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rtp_rtcp_ = RtpRtcp::Create(rtp_config);
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rtp_rtcp_ = ModuleRtpRtcpImpl2::Create(rtp_config);
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rtp_rtcp_->SetSendingMediaStatus(false);
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rtp_rtcp_->SetRTCPStatus(RtcpMode::kCompound);
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@ -36,6 +36,7 @@ if (rtc_include_tests) {
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"../../../api/task_queue:default_task_queue_factory",
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"../../../modules/audio_mixer:audio_mixer_impl",
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"../../../modules/audio_mixer:audio_mixer_test_utils",
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"../../../modules/rtp_rtcp:rtp_rtcp",
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"../../../modules/rtp_rtcp:rtp_rtcp_format",
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"../../../modules/utility",
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"../../../rtc_base:logging",
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@ -56,6 +57,7 @@ if (rtc_include_tests) {
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"../../../api/audio_codecs:builtin_audio_encoder_factory",
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"../../../api/task_queue:default_task_queue_factory",
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"../../../modules/audio_mixer:audio_mixer_test_utils",
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"../../../modules/rtp_rtcp:rtp_rtcp",
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"../../../rtc_base:logging",
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"../../../rtc_base:rtc_event",
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"../../../test:mock_transport",
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@ -72,6 +74,7 @@ if (rtc_include_tests) {
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"../../../api/audio_codecs:builtin_audio_encoder_factory",
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"../../../api/task_queue:default_task_queue_factory",
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"../../../modules/audio_mixer:audio_mixer_test_utils",
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"../../../modules/rtp_rtcp:rtp_rtcp",
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"../../../modules/rtp_rtcp:rtp_rtcp_format",
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"../../../rtc_base:logging",
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"../../../rtc_base:rtc_event",
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@ -14,6 +14,7 @@
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#include "api/task_queue/default_task_queue_factory.h"
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#include "modules/audio_mixer/sine_wave_generator.h"
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#include "modules/rtp_rtcp/source/rtp_packet_received.h"
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#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
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#include "rtc_base/event.h"
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#include "rtc_base/logging.h"
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#include "test/gmock.h"
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@ -36,7 +37,7 @@ std::unique_ptr<RtpRtcp> CreateRtpStack(Clock* clock,
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rtp_config.rtcp_report_interval_ms = 5000;
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rtp_config.outgoing_transport = transport;
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rtp_config.local_media_ssrc = remote_ssrc;
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auto rtp_rtcp = RtpRtcp::Create(rtp_config);
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auto rtp_rtcp = ModuleRtpRtcpImpl2::Create(rtp_config);
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rtp_rtcp->SetSendingMediaStatus(false);
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rtp_rtcp->SetRTCPStatus(RtcpMode::kCompound);
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return rtp_rtcp;
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@ -15,6 +15,7 @@
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#include "api/task_queue/default_task_queue_factory.h"
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#include "audio/voip/audio_egress.h"
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#include "modules/audio_mixer/sine_wave_generator.h"
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#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
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#include "rtc_base/event.h"
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#include "rtc_base/logging.h"
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#include "test/gmock.h"
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@ -45,7 +46,7 @@ class AudioIngressTest : public ::testing::Test {
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rtp_config.rtcp_report_interval_ms = 5000;
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rtp_config.outgoing_transport = &transport_;
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rtp_config.local_media_ssrc = 0xdeadc0de;
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rtp_rtcp_ = RtpRtcp::Create(rtp_config);
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rtp_rtcp_ = ModuleRtpRtcpImpl2::Create(rtp_config);
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rtp_rtcp_->SetSendingMediaStatus(false);
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rtp_rtcp_->SetRTCPStatus(RtcpMode::kCompound);
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@ -22,8 +22,8 @@
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#include "call/rtp_stream_receiver_controller_interface.h"
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#include "modules/rtp_rtcp/include/flexfec_receiver.h"
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#include "modules/rtp_rtcp/include/receive_statistics.h"
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#include "modules/rtp_rtcp/include/rtp_rtcp.h"
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#include "modules/rtp_rtcp/source/rtp_packet_received.h"
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#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
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#include "modules/utility/include/process_thread.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/location.h"
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@ -132,7 +132,7 @@ std::unique_ptr<RtpRtcp> CreateRtpRtcpModule(
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configuration.outgoing_transport = config.rtcp_send_transport;
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configuration.rtt_stats = rtt_stats;
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configuration.local_media_ssrc = config.local_ssrc;
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return RtpRtcp::Create(configuration);
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return ModuleRtpRtcpImpl2::Create(configuration);
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}
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} // namespace
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@ -22,8 +22,8 @@
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#include "api/video_codecs/video_codec.h"
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#include "call/rtp_transport_controller_send_interface.h"
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#include "modules/pacing/packet_router.h"
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#include "modules/rtp_rtcp/include/rtp_rtcp.h"
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#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
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#include "modules/rtp_rtcp/source/rtp_sender.h"
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#include "modules/utility/include/process_thread.h"
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#include "modules/video_coding/include/video_codec_interface.h"
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@ -253,7 +253,7 @@ std::vector<RtpStreamSender> CreateRtpStreamSenders(
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configuration.need_rtp_packet_infos = rtp_config.lntf.enabled;
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auto rtp_rtcp = RtpRtcp::Create(configuration);
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auto rtp_rtcp = ModuleRtpRtcpImpl2::Create(configuration);
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rtp_rtcp->SetSendingStatus(false);
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rtp_rtcp->SetSendingMediaStatus(false);
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rtp_rtcp->SetRTCPStatus(RtcpMode::kCompound);
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@ -50,6 +50,7 @@ namespace rtcp {
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class TransportFeedback;
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}
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// TODO(tommi): See if we can remove Module.
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class RtpRtcp : public Module, public RtcpFeedbackSenderInterface {
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public:
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struct Configuration {
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@ -158,8 +159,15 @@ class RtpRtcp : public Module, public RtcpFeedbackSenderInterface {
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RTC_DISALLOW_COPY_AND_ASSIGN(Configuration);
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};
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// Creates an RTP/RTCP module object using provided |configuration|.
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static std::unique_ptr<RtpRtcp> Create(const Configuration& configuration);
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// DEPRECATED. Do not use. Currently instantiates a deprecated version of the
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// RtpRtcp module.
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static std::unique_ptr<RtpRtcp> RTC_DEPRECATED
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Create(const Configuration& configuration) {
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return DEPRECATED_Create(configuration);
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}
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static std::unique_ptr<RtpRtcp> DEPRECATED_Create(
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const Configuration& configuration);
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// **************************************************************************
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// Receiver functions
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@ -19,9 +19,9 @@
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#include "call/rtp_stream_receiver_controller.h"
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#include "call/rtx_receive_stream.h"
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#include "modules/rtp_rtcp/include/receive_statistics.h"
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#include "modules/rtp_rtcp/include/rtp_rtcp.h"
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#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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#include "modules/rtp_rtcp/source/rtp_packet_received.h"
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#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
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#include "modules/rtp_rtcp/source/rtp_sender_video.h"
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#include "rtc_base/rate_limiter.h"
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#include "test/gtest.h"
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@ -134,7 +134,7 @@ class RtpRtcpRtxNackTest : public ::testing::Test {
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configuration.retransmission_rate_limiter = &retransmission_rate_limiter_;
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configuration.local_media_ssrc = kTestSsrc;
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configuration.rtx_send_ssrc = kTestRtxSsrc;
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rtp_rtcp_module_ = RtpRtcp::Create(configuration);
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rtp_rtcp_module_ = ModuleRtpRtcpImpl2::Create(configuration);
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FieldTrialBasedConfig field_trials;
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RTPSenderVideo::Config video_config;
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video_config.clock = &fake_clock;
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@ -48,6 +48,14 @@ ModuleRtpRtcpImpl::RtpSenderContext::RtpSenderContext(
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&packet_history,
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config.paced_sender ? config.paced_sender : &non_paced_sender) {}
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std::unique_ptr<RtpRtcp> RtpRtcp::DEPRECATED_Create(
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const Configuration& configuration) {
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RTC_DCHECK(configuration.clock);
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RTC_LOG(LS_ERROR)
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<< "*********** USING WebRTC INTERNAL IMPLEMENTATION DETAILS ***********";
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return std::make_unique<ModuleRtpRtcpImpl>(configuration);
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}
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ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration)
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: rtcp_sender_(configuration),
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rtcp_receiver_(configuration, this),
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@ -48,11 +48,6 @@ ModuleRtpRtcpImpl2::RtpSenderContext::RtpSenderContext(
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&packet_history,
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config.paced_sender ? config.paced_sender : &non_paced_sender) {}
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std::unique_ptr<RtpRtcp> RtpRtcp::Create(const Configuration& configuration) {
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RTC_DCHECK(configuration.clock);
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return std::make_unique<ModuleRtpRtcpImpl2>(configuration);
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}
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ModuleRtpRtcpImpl2::ModuleRtpRtcpImpl2(const Configuration& configuration)
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: rtcp_sender_(configuration),
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rtcp_receiver_(configuration, this),
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@ -86,6 +81,14 @@ ModuleRtpRtcpImpl2::~ModuleRtpRtcpImpl2() {
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RTC_DCHECK_RUN_ON(&construction_thread_checker_);
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}
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// static
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std::unique_ptr<RtpRtcp> ModuleRtpRtcpImpl2::Create(
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const Configuration& configuration) {
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RTC_DCHECK(configuration.clock);
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RTC_DCHECK(TaskQueueBase::Current());
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return std::make_unique<ModuleRtpRtcpImpl2>(configuration);
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}
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// Returns the number of milliseconds until the module want a worker thread
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// to call Process.
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int64_t ModuleRtpRtcpImpl2::TimeUntilNextProcess() {
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@ -24,13 +24,13 @@
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#include "api/video/video_bitrate_allocation.h"
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#include "modules/include/module_fec_types.h"
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#include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
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#include "modules/rtp_rtcp/include/rtp_rtcp.h"
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#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" // RTCPPacketType
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#include "modules/rtp_rtcp/source/rtcp_packet/tmmb_item.h"
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#include "modules/rtp_rtcp/source/rtcp_receiver.h"
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#include "modules/rtp_rtcp/source/rtcp_sender.h"
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#include "modules/rtp_rtcp/source/rtp_packet_history.h"
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#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
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#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
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#include "modules/rtp_rtcp/source/rtp_sender.h"
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#include "modules/rtp_rtcp/source/rtp_sender_egress.h"
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#include "rtc_base/critical_section.h"
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@ -49,6 +49,14 @@ class ModuleRtpRtcpImpl2 final : public RtpRtcp,
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explicit ModuleRtpRtcpImpl2(const RtpRtcp::Configuration& configuration);
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~ModuleRtpRtcpImpl2() override;
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// This method is provided to easy with migrating away from the
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// RtpRtcp::Create factory method. Since this is an internal implementation
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// detail though, creating an instance of ModuleRtpRtcpImpl2 directly should
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// be fine.
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static std::unique_ptr<RtpRtcp> Create(const Configuration& configuration);
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// TODO(tommi): Make implementation private?
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// Returns the number of milliseconds until the module want a worker thread to
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// call Process.
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int64_t TimeUntilNextProcess() override;
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@ -18,6 +18,7 @@
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#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
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#include "modules/rtp_rtcp/source/rtp_packet_received.h"
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#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
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#include "modules/rtp_rtcp/source/time_util.h"
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#include "test/gmock.h"
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#include "test/gtest.h"
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@ -67,7 +68,7 @@ class RtpSenderAudioTest : public ::testing::Test {
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public:
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RtpSenderAudioTest()
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: fake_clock_(kStartTime),
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rtp_module_(RtpRtcp::Create([&] {
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rtp_module_(ModuleRtpRtcpImpl2::Create([&] {
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RtpRtcp::Configuration config;
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config.audio = true;
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config.clock = &fake_clock_;
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@ -24,7 +24,6 @@
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#include "common_video/generic_frame_descriptor/generic_frame_info.h"
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#include "modules/rtp_rtcp/include/rtp_cvo.h"
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#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
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#include "modules/rtp_rtcp/include/rtp_rtcp.h"
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#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
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#include "modules/rtp_rtcp/source/rtp_dependency_descriptor_extension.h"
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#include "modules/rtp_rtcp/source/rtp_descriptor_authentication.h"
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@ -33,6 +32,7 @@
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#include "modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension.h"
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#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
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#include "modules/rtp_rtcp/source/rtp_packet_received.h"
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#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
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#include "modules/rtp_rtcp/source/time_util.h"
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#include "rtc_base/arraysize.h"
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#include "rtc_base/rate_limiter.h"
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@ -169,7 +169,7 @@ class RtpSenderVideoTest : public ::testing::TestWithParam<bool> {
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: field_trials_(GetParam()),
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fake_clock_(kStartTime),
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retransmission_rate_limiter_(&fake_clock_, 1000),
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rtp_module_(RtpRtcp::Create([&] {
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rtp_module_(ModuleRtpRtcpImpl2::Create([&] {
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RtpRtcp::Configuration config;
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config.clock = &fake_clock_;
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config.outgoing_transport = &transport_;
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@ -920,7 +920,7 @@ class RtpSenderVideoWithFrameTransformerTest : public ::testing::Test {
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RtpSenderVideoWithFrameTransformerTest()
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: fake_clock_(kStartTime),
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retransmission_rate_limiter_(&fake_clock_, 1000),
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rtp_module_(RtpRtcp::Create([&] {
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rtp_module_(ModuleRtpRtcpImpl2::Create([&] {
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RtpRtcp::Configuration config;
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config.clock = &fake_clock_;
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config.outgoing_transport = &transport_;
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@ -16,7 +16,7 @@
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#include "api/video/video_bitrate_allocation.h"
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#include "call/fake_network_pipe.h"
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#include "call/simulated_network.h"
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#include "modules/rtp_rtcp/include/rtp_rtcp.h"
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#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
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#include "rtc_base/rate_limiter.h"
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#include "rtc_base/task_queue_for_test.h"
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#include "rtc_base/task_utils/to_queued_task.h"
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@ -244,7 +244,7 @@ TEST_F(BandwidthEndToEndTest, RembWithSendSideBwe) {
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config.outgoing_transport = receive_transport_;
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config.retransmission_rate_limiter = &retransmission_rate_limiter_;
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config.local_media_ssrc = (*receive_configs)[0].rtp.local_ssrc;
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rtp_rtcp_ = RtpRtcp::Create(config);
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rtp_rtcp_ = ModuleRtpRtcpImpl2::Create(config);
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rtp_rtcp_->SetRemoteSSRC((*receive_configs)[0].rtp.remote_ssrc);
|
||||
rtp_rtcp_->SetRTCPStatus(RtcpMode::kReducedSize);
|
||||
}
|
||||
|
||||
@ -25,7 +25,6 @@
|
||||
#include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
|
||||
#include "modules/rtp_rtcp/include/receive_statistics.h"
|
||||
#include "modules/rtp_rtcp/include/rtp_cvo.h"
|
||||
#include "modules/rtp_rtcp/include/rtp_rtcp.h"
|
||||
#include "modules/rtp_rtcp/include/ulpfec_receiver.h"
|
||||
#include "modules/rtp_rtcp/source/create_video_rtp_depacketizer.h"
|
||||
#include "modules/rtp_rtcp/source/rtp_dependency_descriptor_extension.h"
|
||||
@ -35,6 +34,7 @@
|
||||
#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
|
||||
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
|
||||
#include "modules/rtp_rtcp/source/rtp_rtcp_config.h"
|
||||
#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
|
||||
#include "modules/rtp_rtcp/source/video_rtp_depacketizer.h"
|
||||
#include "modules/rtp_rtcp/source/video_rtp_depacketizer_raw.h"
|
||||
#include "modules/utility/include/process_thread.h"
|
||||
@ -97,7 +97,7 @@ std::unique_ptr<RtpRtcp> CreateRtpRtcpModule(
|
||||
configuration.rtcp_cname_callback = rtcp_cname_callback;
|
||||
configuration.local_media_ssrc = local_ssrc;
|
||||
|
||||
std::unique_ptr<RtpRtcp> rtp_rtcp = RtpRtcp::Create(configuration);
|
||||
std::unique_ptr<RtpRtcp> rtp_rtcp = RtpRtcp::DEPRECATED_Create(configuration);
|
||||
rtp_rtcp->SetRTCPStatus(RtcpMode::kCompound);
|
||||
|
||||
return rtp_rtcp;
|
||||
|
||||
@ -25,7 +25,6 @@
|
||||
#include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
|
||||
#include "modules/rtp_rtcp/include/receive_statistics.h"
|
||||
#include "modules/rtp_rtcp/include/rtp_cvo.h"
|
||||
#include "modules/rtp_rtcp/include/rtp_rtcp.h"
|
||||
#include "modules/rtp_rtcp/include/ulpfec_receiver.h"
|
||||
#include "modules/rtp_rtcp/source/create_video_rtp_depacketizer.h"
|
||||
#include "modules/rtp_rtcp/source/rtp_dependency_descriptor_extension.h"
|
||||
@ -35,6 +34,7 @@
|
||||
#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
|
||||
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
|
||||
#include "modules/rtp_rtcp/source/rtp_rtcp_config.h"
|
||||
#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
|
||||
#include "modules/rtp_rtcp/source/video_rtp_depacketizer.h"
|
||||
#include "modules/rtp_rtcp/source/video_rtp_depacketizer_raw.h"
|
||||
#include "modules/utility/include/process_thread.h"
|
||||
@ -97,7 +97,7 @@ std::unique_ptr<RtpRtcp> CreateRtpRtcpModule(
|
||||
configuration.rtcp_cname_callback = rtcp_cname_callback;
|
||||
configuration.local_media_ssrc = local_ssrc;
|
||||
|
||||
std::unique_ptr<RtpRtcp> rtp_rtcp = RtpRtcp::Create(configuration);
|
||||
std::unique_ptr<RtpRtcp> rtp_rtcp = ModuleRtpRtcpImpl2::Create(configuration);
|
||||
rtp_rtcp->SetRTCPStatus(RtcpMode::kCompound);
|
||||
|
||||
return rtp_rtcp;
|
||||
|
||||
@ -37,6 +37,7 @@
|
||||
#include "test/gmock.h"
|
||||
#include "test/gtest.h"
|
||||
#include "test/mock_frame_transformer.h"
|
||||
#include "test/time_controller/simulated_task_queue.h"
|
||||
|
||||
using ::testing::_;
|
||||
using ::testing::ElementsAre;
|
||||
@ -237,6 +238,9 @@ class RtpVideoStreamReceiver2Test : public ::testing::Test {
|
||||
return config;
|
||||
}
|
||||
|
||||
TokenTaskQueue task_queue_;
|
||||
TokenTaskQueue::CurrentTaskQueueSetter task_queue_setter_{&task_queue_};
|
||||
|
||||
const webrtc::test::ScopedFieldTrials override_field_trials_;
|
||||
VideoReceiveStream::Config config_;
|
||||
MockNackSender mock_nack_sender_;
|
||||
|
||||
@ -25,10 +25,10 @@
|
||||
#include "call/simulated_network.h"
|
||||
#include "call/video_send_stream.h"
|
||||
#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
|
||||
#include "modules/rtp_rtcp/include/rtp_rtcp.h"
|
||||
#include "modules/rtp_rtcp/source/rtcp_sender.h"
|
||||
#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
|
||||
#include "modules/rtp_rtcp/source/rtp_packet.h"
|
||||
#include "modules/rtp_rtcp/source/rtp_rtcp_impl2.h"
|
||||
#include "modules/rtp_rtcp/source/video_rtp_depacketizer_vp9.h"
|
||||
#include "modules/video_coding/codecs/vp8/include/vp8.h"
|
||||
#include "modules/video_coding/codecs/vp9/include/vp9.h"
|
||||
@ -1677,7 +1677,7 @@ TEST_F(VideoSendStreamTest, MinTransmitBitrateRespectsRemb) {
|
||||
config.clock = Clock::GetRealTimeClock();
|
||||
config.outgoing_transport = feedback_transport_.get();
|
||||
config.retransmission_rate_limiter = &retranmission_rate_limiter_;
|
||||
rtp_rtcp_ = RtpRtcp::Create(config);
|
||||
rtp_rtcp_ = ModuleRtpRtcpImpl2::Create(config);
|
||||
rtp_rtcp_->SetRTCPStatus(RtcpMode::kReducedSize);
|
||||
}
|
||||
|
||||
|
||||
Loading…
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Reference in New Issue
Block a user